The Internet has evolved into an essential communication tool for millions of users in the business, technical and educational fields. In this regard, a growing use of the Internet relates to Internet telephony which provides a number of advantages over conventional circuit-switched network telephony systems that are controlled by a separate signaling network.
An important feature in most modem telephony systems is multi-party conferencing. Multi-party conferencing can range from simple three party calls to multi-casts involving thousands of participants. Internet telephony systems generally use either the H.323 signaling protocol or the session initiation protocol (SIP) for signaling and call control functions. In the case of H.323, this protocol includes a defined multipoint control unit (MCU) for handling multi-party conferences. Although SIP supports various multi-party conferencing models, there is no rigid definition for a conferencing entity in SIP. In addition, as the H.323 protocol and SIP continue to compete in the market place, it will be increasingly important to provide systems which can effectively establish conferences among users whose equipment is compliant with only one of these two signaling protocols.
The session initiation protocol (SIP) is gaining in popularity as a standard signaling protocol for use in Internet telephony. As this popularity grows, it will be increasingly desirable to provide a system architecture and method for providing improved conferencing services in SIP based systems.